From 15870f05e90a365f8022da416e713be0c5024e2f Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 5 Oct 2009 08:25:13 +0200
Subject: ALSA: hda - Fix invalid initializations for ALC861 auto mode

The recent auto-parser doesn't work for machines with a single output
with ALC861, such as Toshiba laptops, because alc_subsystem_id() sets
the hp_pins[0] while it's listed in line_outs[0].
This ends up with the doubled initialization of the same mixer widget,
and it mutes the DAC route because hp_pins has no DAC assigned.

To fix this problem, just check spec->autocfg.hp_outs and speaker_outs
so that they are really detected pins.

Reference: Novell bnc#544161
	http://bugzilla.novell.com/show_bug.cgi?id=544161

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 15 ++++++++-------
 1 file changed, 8 insertions(+), 7 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 7810d3dcad83..c1e05994cc31 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -14357,15 +14357,16 @@ static void alc861_auto_init_multi_out(struct hda_codec *codec)
 static void alc861_auto_init_hp_out(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	hda_nid_t pin;
 
-	pin = spec->autocfg.hp_pins[0];
-	if (pin)
-		alc861_auto_set_output_and_unmute(codec, pin, PIN_HP,
+	if (spec->autocfg.hp_outs)
+		alc861_auto_set_output_and_unmute(codec,
+						  spec->autocfg.hp_pins[0],
+						  PIN_HP,
 						  spec->multiout.hp_nid);
-	pin = spec->autocfg.speaker_pins[0];
-	if (pin)
-		alc861_auto_set_output_and_unmute(codec, pin, PIN_OUT,
+	if (spec->autocfg.speaker_outs)
+		alc861_auto_set_output_and_unmute(codec,
+						  spec->autocfg.speaker_pins[0],
+						  PIN_OUT,
 						  spec->multiout.dac_nids[0]);
 }
 
-- 
cgit v1.2.3


From f8f25ba3563dab14b1c3ea4d829642b8a61ca5d7 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 6 Oct 2009 08:31:29 +0200
Subject: ALSA: hda - Add a workaround for ASUS A7K

ASUS A7K needs additional GPIO1 bit setup; it has to be cleared.
Added a new fixup hook for this laptop so that it works as is.

Refernece: Novell bnc#494309
	http://bugzilla.novell.com/show_bug.cgi?id=494309

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 59 +++++++++++++++++++++++++++++++++++--------
 1 file changed, 48 insertions(+), 11 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index c1e05994cc31..901c2999ed64 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1362,7 +1362,7 @@ static void alc_ssid_check(struct hda_codec *codec,
 }
 
 /*
- * Fix-up pin default configurations
+ * Fix-up pin default configurations and add default verbs
  */
 
 struct alc_pincfg {
@@ -1370,9 +1370,14 @@ struct alc_pincfg {
 	u32 val;
 };
 
-static void alc_fix_pincfg(struct hda_codec *codec,
+struct alc_fixup {
+	const struct alc_pincfg *pins;
+	const struct hda_verb *verbs;
+};
+
+static void alc_pick_fixup(struct hda_codec *codec,
 			   const struct snd_pci_quirk *quirk,
-			   const struct alc_pincfg **pinfix)
+			   const struct alc_fixup *fix)
 {
 	const struct alc_pincfg *cfg;
 
@@ -1380,9 +1385,14 @@ static void alc_fix_pincfg(struct hda_codec *codec,
 	if (!quirk)
 		return;
 
-	cfg = pinfix[quirk->value];
-	for (; cfg->nid; cfg++)
-		snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val);
+	fix += quirk->value;
+	cfg = fix->pins;
+	if (cfg) {
+		for (; cfg->nid; cfg++)
+			snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val);
+	}
+	if (fix->verbs)
+		add_verb(codec->spec, fix->verbs);
 }
 
 /*
@@ -9593,11 +9603,13 @@ static struct alc_pincfg alc882_abit_aw9d_pinfix[] = {
 	{ }
 };
 
-static const struct alc_pincfg *alc882_pin_fixes[] = {
-	[PINFIX_ABIT_AW9D_MAX] = alc882_abit_aw9d_pinfix,
+static const struct alc_fixup alc882_fixups[] = {
+	[PINFIX_ABIT_AW9D_MAX] = {
+		.pins = alc882_abit_aw9d_pinfix
+	},
 };
 
-static struct snd_pci_quirk alc882_pinfix_tbl[] = {
+static struct snd_pci_quirk alc882_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX),
 	{}
 };
@@ -9869,7 +9881,7 @@ static int patch_alc882(struct hda_codec *codec)
 		board_config = ALC882_AUTO;
 	}
 
-	alc_fix_pincfg(codec, alc882_pinfix_tbl, alc882_pin_fixes);
+	alc_pick_fixup(codec, alc882_fixup_tbl, alc882_fixups);
 
 	if (board_config == ALC882_AUTO) {
 		/* automatic parse from the BIOS config */
@@ -15159,7 +15171,7 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1019, 0xa88d, "Realtek ALC660 demo", ALC660VD_3ST),
 	SND_PCI_QUIRK(0x103c, 0x30bf, "HP TX1000", ALC861VD_HP),
 	SND_PCI_QUIRK(0x1043, 0x12e2, "Asus z35m", ALC660VD_3ST),
-	SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST),
+	/*SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST),*/ /* auto */
 	SND_PCI_QUIRK(0x1043, 0x1633, "Asus V1Sn", ALC660VD_ASUS_V1S),
 	SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST_DIG),
 	SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST),
@@ -15552,6 +15564,29 @@ static void alc861vd_auto_init(struct hda_codec *codec)
 		alc_inithook(codec);
 }
 
+enum {
+	ALC660VD_FIX_ASUS_GPIO1
+};
+
+/* reset GPIO1 */
+static const struct hda_verb alc660vd_fix_asus_gpio1_verbs[] = {
+	{0x01, AC_VERB_SET_GPIO_MASK, 0x03},
+	{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
+	{0x01, AC_VERB_SET_GPIO_DATA, 0x01},
+	{ }
+};
+
+static const struct alc_fixup alc861vd_fixups[] = {
+	[ALC660VD_FIX_ASUS_GPIO1] = {
+		.verbs = alc660vd_fix_asus_gpio1_verbs,
+	},
+};
+
+static struct snd_pci_quirk alc861vd_fixup_tbl[] = {
+	SND_PCI_QUIRK(0x1043, 0x1339, "ASUS A7-K", ALC660VD_FIX_ASUS_GPIO1),
+	{}
+};
+
 static int patch_alc861vd(struct hda_codec *codec)
 {
 	struct alc_spec *spec;
@@ -15573,6 +15608,8 @@ static int patch_alc861vd(struct hda_codec *codec)
 		board_config = ALC861VD_AUTO;
 	}
 
+	alc_pick_fixup(codec, alc861vd_fixup_tbl, alc861vd_fixups);
+
 	if (board_config == ALC861VD_AUTO) {
 		/* automatic parse from the BIOS config */
 		err = alc861vd_parse_auto_config(codec);
-- 
cgit v1.2.3


From 01d4825df62d1d405035b90294bf38616d3f380b Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 6 Oct 2009 13:21:54 +0200
Subject: ALSA: hda - Don't pick up invalid HP pins in alc_subsystem_id()

alc_subsystem_id() tries to pick up a headphone pin if not configured,
but this caused side-effects as the problem in commit
15870f05e90a365f8022da416e713be0c5024e2f.

This patch fixes the driver behavior to pick up invalid HP pins; at least,
the pins that are listed as the primary outputs aren't taken any more.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 11 ++++++++---
 1 file changed, 8 insertions(+), 3 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 901c2999ed64..a61fbbb41b29 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1332,15 +1332,20 @@ do_sku:
 	 *	        when the external headphone out jack is plugged"
 	 */
 	if (!spec->autocfg.hp_pins[0]) {
+		hda_nid_t nid;
 		tmp = (ass >> 11) & 0x3;	/* HP to chassis */
 		if (tmp == 0)
-			spec->autocfg.hp_pins[0] = porta;
+			nid = porta;
 		else if (tmp == 1)
-			spec->autocfg.hp_pins[0] = porte;
+			nid = porte;
 		else if (tmp == 2)
-			spec->autocfg.hp_pins[0] = portd;
+			nid = portd;
 		else
 			return 1;
+		for (i = 0; i < spec->autocfg.line_outs; i++)
+			if (spec->autocfg.line_out_pins[i] == nid)
+				return 1;
+		spec->autocfg.hp_pins[0] = nid;
 	}
 
 	alc_init_auto_hp(codec);
-- 
cgit v1.2.3


From 2fb930b53f513cbc4c102d415d2923a8a7091337 Mon Sep 17 00:00:00 2001
From: Clemens Ladisch <clemens@ladisch.de>
Date: Tue, 6 Oct 2009 08:21:04 +0200
Subject: sound: via82xx: move DXS volume controls to PCM interface

The "VIA DXS" controls are actually volume controls that apply to the
four PCM substreams, so we better indicate this connection by moving the
controls to the PCM interface.

Commit b452e08e73c0e3dbb0be82130217be4b7084299e in 2.6.30 broke the
restoring of these volumes by "alsactl restore" that most distributions
use; the renaming in this patch cures that regression by preventing
alsactl from applying the old, wrong volume levels to the new controls.
http://bugzilla.kernel.org/show_bug.cgi?id=14151
http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=532613

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/via82xx.c | 27 ++++++++++++++++++---------
 1 file changed, 18 insertions(+), 9 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index acfa4760da49..91683a349035 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -1626,7 +1626,7 @@ static int snd_via8233_dxs_volume_get(struct snd_kcontrol *kcontrol,
 				      struct snd_ctl_elem_value *ucontrol)
 {
 	struct via82xx *chip = snd_kcontrol_chip(kcontrol);
-	unsigned int idx = snd_ctl_get_ioff(kcontrol, &ucontrol->id);
+	unsigned int idx = kcontrol->id.subdevice;
 
 	ucontrol->value.integer.value[0] = VIA_DXS_MAX_VOLUME - chip->playback_volume[idx][0];
 	ucontrol->value.integer.value[1] = VIA_DXS_MAX_VOLUME - chip->playback_volume[idx][1];
@@ -1646,7 +1646,7 @@ static int snd_via8233_dxs_volume_put(struct snd_kcontrol *kcontrol,
 				      struct snd_ctl_elem_value *ucontrol)
 {
 	struct via82xx *chip = snd_kcontrol_chip(kcontrol);
-	unsigned int idx = snd_ctl_get_ioff(kcontrol, &ucontrol->id);
+	unsigned int idx = kcontrol->id.subdevice;
 	unsigned long port = chip->port + 0x10 * idx;
 	unsigned char val;
 	int i, change = 0;
@@ -1705,11 +1705,12 @@ static struct snd_kcontrol_new snd_via8233_pcmdxs_volume_control __devinitdata =
 };
 
 static struct snd_kcontrol_new snd_via8233_dxs_volume_control __devinitdata = {
-	.name = "VIA DXS Playback Volume",
-	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.iface = SNDRV_CTL_ELEM_IFACE_PCM,
+	.device = 0,
+	/* .subdevice set later */
+	.name = "PCM Playback Volume",
 	.access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
 		   SNDRV_CTL_ELEM_ACCESS_TLV_READ),
-	.count = 4,
 	.info = snd_via8233_dxs_volume_info,
 	.get = snd_via8233_dxs_volume_get,
 	.put = snd_via8233_dxs_volume_put,
@@ -1936,10 +1937,18 @@ static int __devinit snd_via8233_init_misc(struct via82xx *chip)
 		}
 		else /* Using DXS when PCM emulation is enabled is really weird */
 		{
-			/* Standalone DXS controls */
-			err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_via8233_dxs_volume_control, chip));
-			if (err < 0)
-				return err;
+			for (i = 0; i < 4; ++i) {
+				struct snd_kcontrol *kctl;
+
+				kctl = snd_ctl_new1(
+					&snd_via8233_dxs_volume_control, chip);
+				if (!kctl)
+					return -ENOMEM;
+				kctl->id.subdevice = i;
+				err = snd_ctl_add(chip->card, kctl);
+				if (err < 0)
+					return err;
+			}
 		}
 	}
 	/* select spdif data slot 10/11 */
-- 
cgit v1.2.3


From defb5ab2e0ff08ff9a942e2bb7e14c21a55ec26b Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 7 Oct 2009 15:12:27 +0200
Subject: ALSA: hda - Fix yet another auto-mic bug in ALC268

Since patch_alc268() doesn't call set_capture_mixer() (due to its h/w
design different from other siblings), it needs to call fixup_automic_adc()
explicitly to set up the auto-mic routing.  Otherwise the indices for
int/ext mics aren't set properly.

Reference: Novell bnc#544899
	http://bugzilla.novell.com/show_bug.cgi?id=544899

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 4 +++-
 1 file changed, 3 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a61fbbb41b29..470fd74a0a1a 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -12859,12 +12859,15 @@ static int patch_alc268(struct hda_codec *codec)
 		unsigned int wcap = get_wcaps(codec, 0x07);
 		int i;
 
+		spec->capsrc_nids = alc268_capsrc_nids;
 		/* get type */
 		wcap = get_wcaps_type(wcap);
 		if (spec->auto_mic ||
 		    wcap != AC_WID_AUD_IN || spec->input_mux->num_items == 1) {
 			spec->adc_nids = alc268_adc_nids_alt;
 			spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt);
+			if (spec->auto_mic)
+				fixup_automic_adc(codec);
 			if (spec->auto_mic || spec->input_mux->num_items == 1)
 				add_mixer(spec, alc268_capture_nosrc_mixer);
 			else
@@ -12874,7 +12877,6 @@ static int patch_alc268(struct hda_codec *codec)
 			spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids);
 			add_mixer(spec, alc268_capture_mixer);
 		}
-		spec->capsrc_nids = alc268_capsrc_nids;
 		/* set default input source */
 		for (i = 0; i < spec->num_adc_nids; i++)
 			snd_hda_codec_write_cache(codec, alc268_capsrc_nids[i],
-- 
cgit v1.2.3


From 2bdf66331c3ff8d564efe7a054f1099133d520cd Mon Sep 17 00:00:00 2001
From: Pavel Hofman <pavel.hofman@ivitera.com>
Date: Tue, 6 Oct 2009 16:04:11 +0200
Subject: ALSA: ICE1712/24 - Change the Multi Track Peak control (level meters)
 from MIXER to PCM type

* PLEASE NOTE - this change requires the corresponding update of
  envy24control for ice1712 - kind of an ABI change.
* The "Multi Track Peak" control is read-only level meters indicator.
* The control is VERY confusing to most users since it is currently displayed
  in regular mixers. E.g. alsamixer ignores its read-only status
  and allows changing the levels with keys which makes no sense.

Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ice1712/ice1712.c | 2 +-
 sound/pci/ice1712/ice1724.c | 2 +-
 2 files changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c
index cecf1ffeeaaa..d74033a2cfbe 100644
--- a/sound/pci/ice1712/ice1712.c
+++ b/sound/pci/ice1712/ice1712.c
@@ -2259,7 +2259,7 @@ static int snd_ice1712_pro_peak_get(struct snd_kcontrol *kcontrol,
 }
 
 static struct snd_kcontrol_new snd_ice1712_mixer_pro_peak __devinitdata = {
-	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.iface = SNDRV_CTL_ELEM_IFACE_PCM,
 	.name = "Multi Track Peak",
 	.access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
 	.info = snd_ice1712_pro_peak_info,
diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index af6e00148621..c24f268f63a8 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -2110,7 +2110,7 @@ static int snd_vt1724_pro_peak_get(struct snd_kcontrol *kcontrol,
 }
 
 static struct snd_kcontrol_new snd_vt1724_mixer_pro_peak __devinitdata = {
-	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.iface = SNDRV_CTL_ELEM_IFACE_PCM,
 	.name = "Multi Track Peak",
 	.access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
 	.info = snd_vt1724_pro_peak_info,
-- 
cgit v1.2.3


From 1d4efa6650454177afe30ad97283ff78572d0442 Mon Sep 17 00:00:00 2001
From: Robert Hancock <hancockrwd@gmail.com>
Date: Wed, 7 Oct 2009 20:19:21 -0600
Subject: ALSA: ice1724: increase SPDIF and independent stereo buffer sizes

Increase the default and maximum PCM buffer prellocation size for ice1724's
SPDIF and independent stereo pair outputs to 256K, which is the hardware's
maximum supported size. This allows a reduction in interrupt rate and
potentially power usage when an application is not latency-critical.

Signed-off-by: Robert Hancock <hancockrwd@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ice1712/ice1724.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index c24f268f63a8..76b717dae4b6 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -1294,7 +1294,7 @@ static int __devinit snd_vt1724_pcm_spdif(struct snd_ice1712 *ice, int device)
 
 	snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
 					      snd_dma_pci_data(ice->pci),
-					      64*1024, 64*1024);
+					      256*1024, 256*1024);
 
 	ice->pcm = pcm;
 
@@ -1408,7 +1408,7 @@ static int __devinit snd_vt1724_pcm_indep(struct snd_ice1712 *ice, int device)
 
 	snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
 					      snd_dma_pci_data(ice->pci),
-					      64*1024, 64*1024);
+					      256*1024, 256*1024);
 
 	ice->pcm_ds = pcm;
 
-- 
cgit v1.2.3


From f0613d5752d8f7d1d02e6d40947f38877fdf9c90 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 9 Oct 2009 17:44:08 +0200
Subject: ALSA: hda - Add full rates/formats support for Nvidia HDMI

Allow Nvidia HDMI to support more possible sample rates and formats.
At best, the really supported rates and formats should be determined
together with the negotiation with the HDMI receiver, but it's currently
not implemented yet (Nvidia stuff seems incompatible with HDMI 1.3
standard in this regard).  As a compromise, we enable all bits, assuming
that all recent devices do support such rates/formats.

Tested-by: Alan Alan <alanwww1@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_nvhdmi.c | 31 +++++++++++++++++++++++++------
 1 file changed, 25 insertions(+), 6 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c
index c8435c9a97f9..23ad93983118 100644
--- a/sound/pci/hda/patch_nvhdmi.c
+++ b/sound/pci/hda/patch_nvhdmi.c
@@ -29,6 +29,9 @@
 #include "hda_codec.h"
 #include "hda_local.h"
 
+/* define below to restrict the supported rates and formats */
+#define LIMITED_RATE_FMT_SUPPORT
+
 struct nvhdmi_spec {
 	struct hda_multi_out multiout;
 
@@ -60,6 +63,22 @@ static struct hda_verb nvhdmi_basic_init[] = {
 	{} /* terminator */
 };
 
+#ifdef LIMITED_RATE_FMT_SUPPORT
+/* support only the safe format and rate */
+#define SUPPORTED_RATES		SNDRV_PCM_RATE_48000
+#define SUPPORTED_MAXBPS	16
+#define SUPPORTED_FORMATS	SNDRV_PCM_FMTBIT_S16_LE
+#else
+/* support all rates and formats */
+#define SUPPORTED_RATES \
+	(SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
+	SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 |\
+	 SNDRV_PCM_RATE_192000)
+#define SUPPORTED_MAXBPS	24
+#define SUPPORTED_FORMATS \
+	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE)
+#endif
+
 /*
  * Controls
  */
@@ -258,9 +277,9 @@ static struct hda_pcm_stream nvhdmi_pcm_digital_playback_8ch = {
 	.channels_min = 2,
 	.channels_max = 8,
 	.nid = Nv_Master_Convert_nid,
-	.rates = SNDRV_PCM_RATE_48000,
-	.maxbps = 16,
-	.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	.rates = SUPPORTED_RATES,
+	.maxbps = SUPPORTED_MAXBPS,
+	.formats = SUPPORTED_FORMATS,
 	.ops = {
 		.open = nvhdmi_dig_playback_pcm_open,
 		.close = nvhdmi_dig_playback_pcm_close_8ch,
@@ -273,9 +292,9 @@ static struct hda_pcm_stream nvhdmi_pcm_digital_playback_2ch = {
 	.channels_min = 2,
 	.channels_max = 2,
 	.nid = Nv_Master_Convert_nid,
-	.rates = SNDRV_PCM_RATE_48000,
-	.maxbps = 16,
-	.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	.rates = SUPPORTED_RATES,
+	.maxbps = SUPPORTED_MAXBPS,
+	.formats = SUPPORTED_FORMATS,
 	.ops = {
 		.open = nvhdmi_dig_playback_pcm_open,
 		.close = nvhdmi_dig_playback_pcm_close_2ch,
-- 
cgit v1.2.3


From 43189a38dada053b820fafc47de8ba665dd3a618 Mon Sep 17 00:00:00 2001
From: Robert Hancock <hancockrwd@gmail.com>
Date: Fri, 9 Oct 2009 22:08:58 -0600
Subject: ALSA: ice1724: Fix surround on Chaintech AV-710

Fix the num_total_dacs setting for Chaintech AV710. The existing comment
that only PSDOUT0 is connected is correct, but since the card is using
packed AC97 mode to send 6 channels to the codec, num_total_dacs should be
set to 6 and not 2. This allows 6-channel surround to work. Also clarify
a comment regarding the additional WM8728 codec on this card (it's connected
to the SPDIF output and always receives the same data).

Signed-off-by: Robert Hancock <hancockrwd@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ice1712/amp.c | 8 +++++---
 1 file changed, 5 insertions(+), 3 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/ice1712/amp.c b/sound/pci/ice1712/amp.c
index 37564300b50d..6da21a2bcade 100644
--- a/sound/pci/ice1712/amp.c
+++ b/sound/pci/ice1712/amp.c
@@ -52,11 +52,13 @@ static int __devinit snd_vt1724_amp_init(struct snd_ice1712 *ice)
 
 	/* only use basic functionality for now */
 
-	ice->num_total_dacs = 2;	/* only PSDOUT0 is connected */
+	/* VT1616 6ch codec connected to PSDOUT0 using packed mode */
+	ice->num_total_dacs = 6;
 	ice->num_total_adcs = 2;
 
-	/* Chaintech AV-710 has another codecs, which need initialization */
-	/* initialize WM8728 codec */
+	/* Chaintech AV-710 has another WM8728 codec connected to PSDOUT4
+	   (shared with the SPDIF output). Mixer control for this codec
+	   is not yet supported. */
 	if (ice->eeprom.subvendor == VT1724_SUBDEVICE_AV710) {
 		for (i = 0; i < ARRAY_SIZE(wm_inits); i += 2)
 			wm_put(ice, wm_inits[i], wm_inits[i+1]);
-- 
cgit v1.2.3


From bd3c200e6d5495343c91db66d2acf1853b57a141 Mon Sep 17 00:00:00 2001
From: David Henningsson <launchpad.web@epost.diwic.se>
Date: Sun, 11 Oct 2009 11:37:22 +0200
Subject: ALSA: ice1724 - Make call to set hw params succeed on ESI Juli@

If two streams are started immediately after one another (such as a
playback and a recording stream), the call to set hw params fails with
EBUSY. This patch makes the call succeed, so playback and recording will
work properly.

Signed-off-by: David Henningsson <launchpad.web@epost.diwic.se>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ice1712/ice1724.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index 76b717dae4b6..10fc92c05574 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -648,7 +648,7 @@ static int snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate,
 	    (inb(ICEMT1724(ice, DMA_PAUSE)) & DMA_PAUSES)) {
 		/* running? we cannot change the rate now... */
 		spin_unlock_irqrestore(&ice->reg_lock, flags);
-		return -EBUSY;
+		return ((rate == ice->cur_rate) && !force) ? 0 : -EBUSY;
 	}
 	if (!force && is_pro_rate_locked(ice)) {
 		spin_unlock_irqrestore(&ice->reg_lock, flags);
-- 
cgit v1.2.3


From 2d9c648295d7bc376305337d29f540a5e411f632 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 13 Oct 2009 08:06:55 +0200
Subject: ALSA: hda - Fix overflow of spec->init_verbs in patch_realtek.c

ALC861-VD lenovo model causes overflow of spec->init_verbs entries due to
the recent changes.  Simply increase the array size to avoid the overflow.

Reported-by: Luca Tettamanti <kronos.it@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 470fd74a0a1a..c08ca660daba 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -275,7 +275,7 @@ struct alc_spec {
 	struct snd_kcontrol_new *cap_mixer;	/* capture mixer */
 	unsigned int beep_amp;	/* beep amp value, set via set_beep_amp() */
 
-	const struct hda_verb *init_verbs[5];	/* initialization verbs
+	const struct hda_verb *init_verbs[10];	/* initialization verbs
 						 * don't forget NULL
 						 * termination!
 						 */
-- 
cgit v1.2.3


From 9c6b8dcefe9a39f36ba11bdd523c0ac5246514c9 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 13 Oct 2009 09:34:28 +0200
Subject: ALSA: bt87x - Add a whitelist for Pinnacle PCTV (11bd:0012)

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/bt87x.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c
index 24585c6c6d01..4e2b925a94cc 100644
--- a/sound/pci/bt87x.c
+++ b/sound/pci/bt87x.c
@@ -808,6 +808,8 @@ static struct pci_device_id snd_bt87x_ids[] = {
 	BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1002, 0x0001, GENERIC),
 	/* Leadtek Winfast tv 2000xp delux */
 	BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x107d, 0x6606, GENERIC),
+	/* Pinnacle PCTV */
+	BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x11bd, 0x0012, GENERIC),
 	/* Voodoo TV 200 */
 	BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x121a, 0x3000, GENERIC),
 	/* Askey Computer Corp. MagicTView'99 */
-- 
cgit v1.2.3


From 54930531a00af5a1c33361a02e67dd1802110465 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Sun, 11 Oct 2009 17:38:29 +0200
Subject: ALSA: hda - Fix mute sound with STAC9227/9228 codecs

On FSC laptops, the sound gets muted gradually when the volume is chnaged.
This is due to the wrong volume-knob widget setup.  The delta bit (bit 7)
shouldn't be set for these devices.

This patch adds a new quirk to set the value 0x7f to the widget 0x24
instead of 0xff.

Reference: Novell bnc#546006
	http://bugzilla.novell.com/show_bug.cgi?id=546006

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 17 +++++++++++++++++
 1 file changed, 17 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index a9b26828a651..75736827425d 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -158,6 +158,7 @@ enum {
 	STAC_D965_5ST_NO_FP,
 	STAC_DELL_3ST,
 	STAC_DELL_BIOS,
+	STAC_927X_VOLKNOB,
 	STAC_927X_MODELS
 };
 
@@ -915,6 +916,14 @@ static struct hda_verb stac927x_core_init[] = {
 	{}
 };
 
+static struct hda_verb stac927x_volknob_core_init[] = {
+	/* don't set delta bit */
+	{0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0x7f},
+	/* enable analog pc beep path */
+	{0x01, AC_VERB_SET_DIGI_CONVERT_2, 1 << 5},
+	{}
+};
+
 static struct hda_verb stac9205_core_init[] = {
 	/* set master volume and direct control */	
 	{ 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
@@ -1999,6 +2008,7 @@ static unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = {
 	[STAC_D965_5ST_NO_FP]  = d965_5st_no_fp_pin_configs,
 	[STAC_DELL_3ST]  = dell_3st_pin_configs,
 	[STAC_DELL_BIOS] = NULL,
+	[STAC_927X_VOLKNOB] = NULL,
 };
 
 static const char *stac927x_models[STAC_927X_MODELS] = {
@@ -2010,6 +2020,7 @@ static const char *stac927x_models[STAC_927X_MODELS] = {
 	[STAC_D965_5ST_NO_FP]	= "5stack-no-fp",
 	[STAC_DELL_3ST]		= "dell-3stack",
 	[STAC_DELL_BIOS]	= "dell-bios",
+	[STAC_927X_VOLKNOB]	= "volknob",
 };
 
 static struct snd_pci_quirk stac927x_cfg_tbl[] = {
@@ -2045,6 +2056,8 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = {
 			   "Intel D965", STAC_D965_5ST),
 	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_INTEL, 0xff00, 0x2500,
 			   "Intel D965", STAC_D965_5ST),
+	/* volume-knob fixes */
+	SND_PCI_QUIRK_VENDOR(0x10cf, "FSC", STAC_927X_VOLKNOB),
 	{} /* terminator */
 };
 
@@ -5616,6 +5629,10 @@ static int patch_stac927x(struct hda_codec *codec)
 		spec->dmux_nids = stac927x_dmux_nids;
 		spec->num_dmuxes = ARRAY_SIZE(stac927x_dmux_nids);
 		break;
+	case STAC_927X_VOLKNOB:
+		spec->num_dmics = 0;
+		spec->init = stac927x_volknob_core_init;
+		break;
 	default:
 		spec->num_dmics = 0;
 		spec->init = stac927x_core_init;
-- 
cgit v1.2.3


From ccca7cdc1b8dd2e7b67e9289a6abf117b11cbe6b Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 13 Oct 2009 15:32:21 +0200
Subject: ALSA: hda - Fix volume-knob setup for Dell laptops with STAC9228

The volume-knob widget needs to be set with 0x7f instead of 0xff
for Dell laptops with STAC9228 codec, too, like the previous commit.

Reference: Novell bnc#545013
	http://bugzilla.novell.com/show_bug.cgi?id=545013

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 12 +++++++++++-
 1 file changed, 11 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 75736827425d..66c0876bf734 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -908,6 +908,16 @@ static struct hda_verb d965_core_init[] = {
 	{}
 };
 
+static struct hda_verb dell_3st_core_init[] = {
+	/* don't set delta bit */
+	{0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0x7f},
+	/* unmute node 0x1b */
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+	/* select node 0x03 as DAC */
+	{0x0b, AC_VERB_SET_CONNECT_SEL, 0x01},
+	{}
+};
+
 static struct hda_verb stac927x_core_init[] = {
 	/* set master volume and direct control */	
 	{ 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
@@ -5625,7 +5635,7 @@ static int patch_stac927x(struct hda_codec *codec)
 		spec->dmic_nids = stac927x_dmic_nids;
 		spec->num_dmics = STAC927X_NUM_DMICS;
 
-		spec->init = d965_core_init;
+		spec->init = dell_3st_core_init;
 		spec->dmux_nids = stac927x_dmux_nids;
 		spec->num_dmuxes = ARRAY_SIZE(stac927x_dmux_nids);
 		break;
-- 
cgit v1.2.3


From 491dc0437d4c56d11f78113eca3953cff87314f3 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 13 Oct 2009 16:07:59 +0200
Subject: ALSA: hda - Allow all formats as default for Nvidia HDMI

In the commit f0613d5752d8f7d1d02e6d40947f38877fdf9c90
    ALSA: hda - Add full rates/formats support for Nvidia HDMI
the flag LIMITIED_RATE_FMT_SUPPORT was set as default, as I forgot
to clear before commit.

Let's enable all formats/rates as default.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_nvhdmi.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c
index 23ad93983118..9fb60276f5c9 100644
--- a/sound/pci/hda/patch_nvhdmi.c
+++ b/sound/pci/hda/patch_nvhdmi.c
@@ -30,7 +30,7 @@
 #include "hda_local.h"
 
 /* define below to restrict the supported rates and formats */
-#define LIMITED_RATE_FMT_SUPPORT
+/* #define LIMITED_RATE_FMT_SUPPORT */
 
 struct nvhdmi_spec {
 	struct hda_multi_out multiout;
-- 
cgit v1.2.3


From 4b7348a15972274eb16182d10987f69da3e95719 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 14 Oct 2009 18:25:23 +0200
Subject: ALSA: hda - Fix capture source checks for ALC662/663 codecs

The ALC662/663 parser calls wrongly alc880_auto_create_input_ctls()
to check the capture source selections.  This should be alc882, instead.

Reference: Novell bnc#546918
	http://bugzilla.novell.com/show_bug.cgi?id=546918

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index c08ca660daba..9b1cff83497f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -17374,7 +17374,7 @@ static int alc662_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin,
 
 /* create playback/capture controls for input pins */
 #define alc662_auto_create_input_ctls \
-	alc880_auto_create_input_ctls
+	alc882_auto_create_input_ctls
 
 static void alc662_auto_set_output_and_unmute(struct hda_codec *codec,
 					      hda_nid_t nid, int pin_type,
-- 
cgit v1.2.3


From 97609458ce972180172ae2cec0483451820e6a41 Mon Sep 17 00:00:00 2001
From: Wu Zhangjin <wuzhangjin@gmail.com>
Date: Thu, 15 Oct 2009 10:22:54 +0800
Subject: ALSA: SND_CS5535AUDIO: Remove the X86 platform dependency

SND_CS5535AUDIO is available on Loongson(MIPS compatible) family
machines, and checked it with ARCH=x86_64, no relative compiling
warnings & errors, so, remove the platform dependency directly.

Reported-by: rixed@happyleptic.org
Acked-by: Andres Salomon <dilinger@collabora.co.uk>
Signed-off-by: Wu Zhangjin <wuzhangjin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/Kconfig | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index fb5ee3cc3968..75c602b5b132 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -259,7 +259,6 @@ config SND_CS5530
 
 config SND_CS5535AUDIO
 	tristate "CS5535/CS5536 Audio"
-	depends on X86 && !X86_64
 	select SND_PCM
 	select SND_AC97_CODEC
 	help
-- 
cgit v1.2.3


From db32f99816f7cbe61c1f75c1560655a3bf52488a Mon Sep 17 00:00:00 2001
From: peer chen <peerchen@gmail.com>
Date: Thu, 15 Oct 2009 16:37:47 +0800
Subject: ALSA: hda_intel: Add the Linux device ID for NVIDIA HDA controller

Add the generic device ID for NVIDIA HDA controller.

Signed-off-by: Peer Chen <peerchen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index c9ad182e1b4b..e340792f6cb3 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2674,6 +2674,7 @@ static struct pci_device_id azx_ids[] = {
 	{ PCI_DEVICE(0x10de, 0x044b), .driver_data = AZX_DRIVER_NVIDIA },
 	{ PCI_DEVICE(0x10de, 0x055c), .driver_data = AZX_DRIVER_NVIDIA },
 	{ PCI_DEVICE(0x10de, 0x055d), .driver_data = AZX_DRIVER_NVIDIA },
+	{ PCI_DEVICE(0x10de, 0x0590), .driver_data = AZX_DRIVER_NVIDIA },
 	{ PCI_DEVICE(0x10de, 0x0774), .driver_data = AZX_DRIVER_NVIDIA },
 	{ PCI_DEVICE(0x10de, 0x0775), .driver_data = AZX_DRIVER_NVIDIA },
 	{ PCI_DEVICE(0x10de, 0x0776), .driver_data = AZX_DRIVER_NVIDIA },
-- 
cgit v1.2.3


From 4b3be6afa4ab8b3fdce39df68bad71f8b85164de Mon Sep 17 00:00:00 2001
From: Julia Lawall <julia@diku.dk>
Date: Sat, 17 Oct 2009 08:33:22 +0200
Subject: ALSA: sound: Move dereference after NULL test and drop unnecessary
 NULL tests

In pcm.c, if the NULL test on pcm is needed, then the dereference should be
after the NULL test.

In dummy.c and ali5451.c, the context of the calls to
snd_card_dummy_new_mixer and snd_ali_free_voice show that dummy and pvoice,
respectively cannot be NULL.

A simplified version of the semantic match that detects this problem is as
follows (http://coccinelle.lip6.fr/):

// <smpl>
@match exists@
expression x, E;
identifier fld;
@@

* x->fld
  ... when != \(x = E\|&x\)
* x == NULL
// </smpl>

Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ali5451/ali5451.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c
index b458d208720b..aaf4da68969c 100644
--- a/sound/pci/ali5451/ali5451.c
+++ b/sound/pci/ali5451/ali5451.c
@@ -973,7 +973,7 @@ static void snd_ali_free_voice(struct snd_ali * codec,
 	void *private_data;
 
 	snd_ali_printk("free_voice: channel=%d\n",pvoice->number);
-	if (pvoice == NULL || !pvoice->use)
+	if (!pvoice->use)
 		return;
 	snd_ali_clear_voices(codec, pvoice->number, pvoice->number);
 	spin_lock_irq(&codec->voice_alloc);
-- 
cgit v1.2.3


From 3d00941371a765779c4e3509214c7e5793cce1fe Mon Sep 17 00:00:00 2001
From: Clemens Ladisch <clemens@ladisch.de>
Date: Thu, 22 Oct 2009 09:04:09 +0200
Subject: sound: via82xx: deactivate DXS controls of inactive streams

Activate the DXS volume controls only when the corresponding stream is
being used.  This makes the behaviour consistent with the other drivers
that have per-stream volume controls.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/via82xx.c | 59 ++++++++++++++++++++++++++++++++++++++++++++++-------
 1 file changed, 52 insertions(+), 7 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index 91683a349035..8a332d2f615c 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -386,6 +386,7 @@ struct via82xx {
 
 	struct snd_pcm *pcms[2];
 	struct snd_rawmidi *rmidi;
+	struct snd_kcontrol *dxs_controls[4];
 
 	struct snd_ac97_bus *ac97_bus;
 	struct snd_ac97 *ac97;
@@ -1216,9 +1217,9 @@ static int snd_via82xx_pcm_open(struct via82xx *chip, struct viadev *viadev,
 
 
 /*
- * open callback for playback on via686 and via823x DSX
+ * open callback for playback on via686
  */
-static int snd_via82xx_playback_open(struct snd_pcm_substream *substream)
+static int snd_via686_playback_open(struct snd_pcm_substream *substream)
 {
 	struct via82xx *chip = snd_pcm_substream_chip(substream);
 	struct viadev *viadev = &chip->devs[chip->playback_devno + substream->number];
@@ -1229,6 +1230,32 @@ static int snd_via82xx_playback_open(struct snd_pcm_substream *substream)
 	return 0;
 }
 
+/*
+ * open callback for playback on via823x DXS
+ */
+static int snd_via8233_playback_open(struct snd_pcm_substream *substream)
+{
+	struct via82xx *chip = snd_pcm_substream_chip(substream);
+	struct viadev *viadev;
+	unsigned int stream;
+	int err;
+
+	viadev = &chip->devs[chip->playback_devno + substream->number];
+	if ((err = snd_via82xx_pcm_open(chip, viadev, substream)) < 0)
+		return err;
+	stream = viadev->reg_offset / 0x10;
+	if (chip->dxs_controls[stream]) {
+		chip->playback_volume[stream][0] = 0;
+		chip->playback_volume[stream][1] = 0;
+		chip->dxs_controls[stream]->vd[0].access &=
+			~SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+		snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE |
+			       SNDRV_CTL_EVENT_MASK_INFO,
+			       &chip->dxs_controls[stream]->id);
+	}
+	return 0;
+}
+
 /*
  * open callback for playback on via823x multi-channel
  */
@@ -1302,10 +1329,26 @@ static int snd_via82xx_pcm_close(struct snd_pcm_substream *substream)
 	return 0;
 }
 
+static int snd_via8233_playback_close(struct snd_pcm_substream *substream)
+{
+	struct via82xx *chip = snd_pcm_substream_chip(substream);
+	struct viadev *viadev = substream->runtime->private_data;
+	unsigned int stream;
+
+	stream = viadev->reg_offset / 0x10;
+	if (chip->dxs_controls[stream]) {
+		chip->dxs_controls[stream]->vd[0].access |=
+			SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+		snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_INFO,
+			       &chip->dxs_controls[stream]->id);
+	}
+	return snd_via82xx_pcm_close(substream);
+}
+
 
 /* via686 playback callbacks */
 static struct snd_pcm_ops snd_via686_playback_ops = {
-	.open =		snd_via82xx_playback_open,
+	.open =		snd_via686_playback_open,
 	.close =	snd_via82xx_pcm_close,
 	.ioctl =	snd_pcm_lib_ioctl,
 	.hw_params =	snd_via82xx_hw_params,
@@ -1331,8 +1374,8 @@ static struct snd_pcm_ops snd_via686_capture_ops = {
 
 /* via823x DSX playback callbacks */
 static struct snd_pcm_ops snd_via8233_playback_ops = {
-	.open =		snd_via82xx_playback_open,
-	.close =	snd_via82xx_pcm_close,
+	.open =		snd_via8233_playback_open,
+	.close =	snd_via8233_playback_close,
 	.ioctl =	snd_pcm_lib_ioctl,
 	.hw_params =	snd_via82xx_hw_params,
 	.hw_free =	snd_via82xx_hw_free,
@@ -1709,8 +1752,9 @@ static struct snd_kcontrol_new snd_via8233_dxs_volume_control __devinitdata = {
 	.device = 0,
 	/* .subdevice set later */
 	.name = "PCM Playback Volume",
-	.access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
-		   SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+		  SNDRV_CTL_ELEM_ACCESS_TLV_READ |
+		  SNDRV_CTL_ELEM_ACCESS_INACTIVE,
 	.info = snd_via8233_dxs_volume_info,
 	.get = snd_via8233_dxs_volume_get,
 	.put = snd_via8233_dxs_volume_put,
@@ -1948,6 +1992,7 @@ static int __devinit snd_via8233_init_misc(struct via82xx *chip)
 				err = snd_ctl_add(chip->card, kctl);
 				if (err < 0)
 					return err;
+				chip->dxs_controls[i] = kctl;
 			}
 		}
 	}
-- 
cgit v1.2.3


From a1bf808849f25a4d668f81415ecebb2da9fecf8e Mon Sep 17 00:00:00 2001
From: Daniel T Chen <crimsun@ubuntu.com>
Date: Sun, 1 Nov 2009 18:32:29 -0500
Subject: ALSA: hda: Use quirk mask for Dell Inspiron Mini9/Vostro A90 using
 ALC268

BugLink: https://bugs.launchpad.net/bugs/368629

We should use a quirk mask for these Dell Inspiron Mini9s and Vostro
A90s, as the model=dell quirk appears to enable audio on them.

Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 9b1cff83497f..148734d16132 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -12602,7 +12602,8 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x015b, "Acer Aspire One",
 						ALC268_ACER_ASPIRE_ONE),
 	SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL),
-	SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron Mini9", ALC268_DELL),
+	SND_PCI_QUIRK_MASK(0x1028, 0xfff0, 0x02b0,
+			"Dell Inspiron Mini9/Vostro A90", ALC268_DELL),
 	/* almost compatible with toshiba but with optional digital outs;
 	 * auto-probing seems working fine
 	 */
-- 
cgit v1.2.3


From ad87c64f00e01a694bf90bddc2b4a6c90796d13c Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 2 Nov 2009 14:23:15 +0100
Subject: ALSA: hda - Don't check invalid HP pin

alc_automute_pin() might be called even if any HP pin is defined, and
it will result in verbs with NID=0.

This patch adds a check for the validity of HP widget before issuing
any verbs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 148734d16132..ff20048504b6 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -965,6 +965,8 @@ static void alc_automute_pin(struct hda_codec *codec)
 	unsigned int nid = spec->autocfg.hp_pins[0];
 	int i;
 
+	if (!nid)
+		return;
 	pincap = snd_hda_query_pin_caps(codec, nid);
 	if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */
 		snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0);
-- 
cgit v1.2.3


From 5bdaaada16363d64e10ae081755d1a8d392429f2 Mon Sep 17 00:00:00 2001
From: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Date: Wed, 4 Nov 2009 07:57:45 +0100
Subject: ALSA: hda - Enable GPIO control for mute LED on HP systems

This patch enables GPIO to control mute LED indicator on the HP systems
with the special string in BIOS and applies it with the correct polarity on
HP B-series systems.

It also restores configuration of the pin intended as the second Headphone
on HP B-series systems but configured as something else in the BIOS to
pass MS DTM.

Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 68 ++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 68 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 66c0876bf734..b513eba2d2f7 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -28,6 +28,7 @@
 #include <linux/delay.h>
 #include <linux/slab.h>
 #include <linux/pci.h>
+#include <linux/dmi.h>
 #include <sound/core.h>
 #include <sound/asoundef.h>
 #include <sound/jack.h>
@@ -1693,6 +1694,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
 		      "DFI LanParty", STAC_92HD71BXX_REF),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fb,
 		      "HP dv4-1222nr", STAC_HP_DV4_1222NR),
+	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x1720,
+			  "HP", STAC_HP_DV5),
 	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3080,
 		      "HP", STAC_HP_DV5),
 	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x30f0,
@@ -4730,6 +4733,26 @@ static int stac92xx_resume(struct hda_codec *codec)
 	return 0;
 }
 
+static int hp_bseries_system(u32 subsystem_id)
+{
+	switch (subsystem_id) {
+	case 0x103c307e:
+	case 0x103c307f:
+	case 0x103c3080:
+	case 0x103c3081:
+	case 0x103c1722:
+	case 0x103c1723:
+	case 0x103c1724:
+	case 0x103c1725:
+	case 0x103c1726:
+	case 0x103c1727:
+	case 0x103c1728:
+	case 0x103c1729:
+		return 1;
+	}
+	return 0;
+}
+
 /*
  * using power check for controlling mute led of HP notebooks
  * check for mute state only on Speakers (nid = 0x10)
@@ -4754,6 +4777,11 @@ static int stac92xx_hp_check_power_status(struct hda_codec *codec,
 		else
 			spec->gpio_data |= spec->gpio_led; /* white */
 
+		if (hp_bseries_system(codec->subsystem_id)) {
+			/* LED state is inverted on these systems */
+			spec->gpio_data ^= spec->gpio_led;
+		}
+
 		stac_gpio_set(codec, spec->gpio_mask,
 			      spec->gpio_dir,
 			      spec->gpio_data);
@@ -5243,6 +5271,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec;
 	struct hda_verb *unmute_init = stac92hd71bxx_unmute_core_init;
+	unsigned int pin_cfg;
 	int err = 0;
 
 	spec  = kzalloc(sizeof(*spec), GFP_KERNEL);
@@ -5426,6 +5455,45 @@ again:
 		break;
 	}
 
+	if (hp_bseries_system(codec->subsystem_id)) {
+		pin_cfg = snd_hda_codec_get_pincfg(codec, 0x0f);
+		if (get_defcfg_device(pin_cfg) == AC_JACK_LINE_OUT ||
+			get_defcfg_device(pin_cfg) == AC_JACK_SPEAKER  ||
+			get_defcfg_device(pin_cfg) == AC_JACK_HP_OUT) {
+			/* It was changed in the BIOS to just satisfy MS DTM.
+			 * Lets turn it back into slaved HP
+			 */
+			pin_cfg = (pin_cfg & (~AC_DEFCFG_DEVICE))
+					| (AC_JACK_HP_OUT <<
+						AC_DEFCFG_DEVICE_SHIFT);
+			pin_cfg = (pin_cfg & (~(AC_DEFCFG_DEF_ASSOC
+							| AC_DEFCFG_SEQUENCE)))
+								| 0x1f;
+			snd_hda_codec_set_pincfg(codec, 0x0f, pin_cfg);
+		}
+	}
+
+	if ((codec->subsystem_id >> 16) == PCI_VENDOR_ID_HP) {
+		const struct dmi_device *dev = NULL;
+		while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING,
+					      NULL, dev))) {
+			if (strcmp(dev->name, "HP_Mute_LED_1")) {
+				switch (codec->vendor_id) {
+				case 0x111d7608:
+					spec->gpio_led = 0x01;
+					break;
+				case 0x111d7600:
+				case 0x111d7601:
+				case 0x111d7602:
+				case 0x111d7603:
+					spec->gpio_led = 0x08;
+					break;
+				}
+				break;
+			}
+		}
+	}
+
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	if (spec->gpio_led) {
 		spec->gpio_mask |= spec->gpio_led;
-- 
cgit v1.2.3


From 798a8a15011e88cc63dbbb15728b42572c152093 Mon Sep 17 00:00:00 2001
From: Daniel Drake <dsd@laptop.org>
Date: Wed, 4 Nov 2009 10:11:07 +0000
Subject: ALSA: hda - Add OLPC XO-1.5 PCI ID

The XO-1.5 laptop now has a unique subvendor/subproduct ID, which can
be used to automatically select the correct CXT5066 configuration.

Signed-off-by: Daniel Drake <dsd@laptop.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 3fbbc8c01e70..6479e65858d3 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -2325,6 +2325,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
 		      CXT5066_LAPTOP),
 	SND_PCI_QUIRK(0x1028, 0x02f5, "Dell",
 		      CXT5066_DELL_LAPTOP),
+	SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5),
 	{}
 };
 
-- 
cgit v1.2.3


From 7e6c3989af5baee999ef9a4424e85938cba8d34a Mon Sep 17 00:00:00 2001
From: Daniel T Chen <crimsun@ubuntu.com>
Date: Wed, 4 Nov 2009 21:03:46 -0500
Subject: ALSA: intel8x0: Mute External Amplifier by default for another Sony
 model

BugLink: https://bugs.launchpad.net/bugs/474972

This Sony model needs External Amplifier muted for audible playback, so
make sure we set the inv_eapd quirk.

Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/intel8x0.c | 6 ++++++
 1 file changed, 6 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 754867ed4785..aac20fb4aad2 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -1948,6 +1948,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = {
 		.name = "HP xw4200",	/* AD1981B*/
 		.type = AC97_TUNE_HP_ONLY
 	},
+	{
+		.subvendor = 0x104d,
+		.subdevice = 0x8144,
+		.name = "Sony",
+		.type = AC97_TUNE_INV_EAPD
+	},
 	{
 		.subvendor = 0x104d,
 		.subdevice = 0x8197,
-- 
cgit v1.2.3


From 78987bdc4e41a425ac113c2c51474f0368fe653a Mon Sep 17 00:00:00 2001
From: Randy Dunlap <randy.dunlap@oracle.com>
Date: Thu, 5 Nov 2009 09:22:30 -0800
Subject: ALSA: hda, move hp_bseries_system

Function hp_bseries_system() is always used, outside of
CONFIG_ boundaries/controls, so move it.

sound/pci/hda/patch_sigmatel.c:5458: error: implicit declaration of function 'hp_bseries_system'

Signed-off-by: Randy Dunlap <randy.dunlap@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 40 ++++++++++++++++++++--------------------
 1 file changed, 20 insertions(+), 20 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index b513eba2d2f7..8eb6508cd991 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4668,6 +4668,26 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res)
 	}
 }
 
+static int hp_bseries_system(u32 subsystem_id)
+{
+	switch (subsystem_id) {
+	case 0x103c307e:
+	case 0x103c307f:
+	case 0x103c3080:
+	case 0x103c3081:
+	case 0x103c1722:
+	case 0x103c1723:
+	case 0x103c1724:
+	case 0x103c1725:
+	case 0x103c1726:
+	case 0x103c1727:
+	case 0x103c1728:
+	case 0x103c1729:
+		return 1;
+	}
+	return 0;
+}
+
 #ifdef CONFIG_PROC_FS
 static void stac92hd_proc_hook(struct snd_info_buffer *buffer,
 			       struct hda_codec *codec, hda_nid_t nid)
@@ -4733,26 +4753,6 @@ static int stac92xx_resume(struct hda_codec *codec)
 	return 0;
 }
 
-static int hp_bseries_system(u32 subsystem_id)
-{
-	switch (subsystem_id) {
-	case 0x103c307e:
-	case 0x103c307f:
-	case 0x103c3080:
-	case 0x103c3081:
-	case 0x103c1722:
-	case 0x103c1723:
-	case 0x103c1724:
-	case 0x103c1725:
-	case 0x103c1726:
-	case 0x103c1727:
-	case 0x103c1728:
-	case 0x103c1729:
-		return 1;
-	}
-	return 0;
-}
-
 /*
  * using power check for controlling mute led of HP notebooks
  * check for mute state only on Speakers (nid = 0x10)
-- 
cgit v1.2.3


From 1a6969788ef2d5bc3169eee59def6b267182f136 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Sat, 7 Nov 2009 09:49:04 +0100
Subject: ALSA: hda - Don't initialize CORB/RIRB for single_cmd mode

So far, CORB/RIRB still remains even if the driver is switched to the
single_cmd mode.  The specification says that this should be disabled,
but I hoped this isn't the case; indeed most devices worked together with
CORB/RIRB.

However, Poulsbo (US15W) seems problematic with this setup, and it
requires to disable CORB/RIRB when single_cmd is used.

Now this patch disables CORB/RIRB initialization when the single_cmd
mode is used.  Also the unsolicited event is disabled because it can't
work without RIRB.

Reported-and-tested-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 12 ++++++++----
 1 file changed, 8 insertions(+), 4 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index e340792f6cb3..6517f589d01d 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -722,9 +722,10 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
 		   chip->last_cmd[addr]);
 	chip->single_cmd = 1;
 	bus->response_reset = 0;
-	/* re-initialize CORB/RIRB */
+	/* release CORB/RIRB */
 	azx_free_cmd_io(chip);
-	azx_init_cmd_io(chip);
+	/* disable unsolicited responses */
+	azx_writel(chip, GCTL, azx_readl(chip, GCTL) & ~ICH6_GCTL_UNSOL);
 	return -1;
 }
 
@@ -865,7 +866,9 @@ static int azx_reset(struct azx *chip)
 	}
 
 	/* Accept unsolicited responses */
-	azx_writel(chip, GCTL, azx_readl(chip, GCTL) | ICH6_GCTL_UNSOL);
+	if (!chip->single_cmd)
+		azx_writel(chip, GCTL, azx_readl(chip, GCTL) |
+			   ICH6_GCTL_UNSOL);
 
 	/* detect codecs */
 	if (!chip->codec_mask) {
@@ -980,7 +983,8 @@ static void azx_init_chip(struct azx *chip)
 	azx_int_enable(chip);
 
 	/* initialize the codec command I/O */
-	azx_init_cmd_io(chip);
+	if (!chip->single_cmd)
+		azx_init_cmd_io(chip);
 
 	/* program the position buffer */
 	azx_writel(chip, DPLBASE, (u32)chip->posbuf.addr);
-- 
cgit v1.2.3


From 95491d902b4ed1bfd8f602aada793d74cc85428b Mon Sep 17 00:00:00 2001
From: Daniel T Chen <crimsun@ubuntu.com>
Date: Sun, 8 Nov 2009 19:03:55 -0500
Subject: ALSA: hda: Use model=auto quirk for Sony VAIO VGN-FW170J using ALC262

BugLink: https://bugs.launchpad.net/bugs/478309

The internal microphone on this VAIO model does not work unless the
"auto" quirk is used.

Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index ff20048504b6..9bb4f75ca436 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -11460,6 +11460,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD),
 	SND_PCI_QUIRK(0x104d, 0x9016, "Sony VAIO", ALC262_AUTO), /* dig-only */
 	SND_PCI_QUIRK(0x104d, 0x9025, "Sony VAIO Z21MN", ALC262_TOSHIBA_S06),
+	SND_PCI_QUIRK(0x104d, 0x9035, "Sony VAIO VGN-FW170J", ALC262_AUTO),
 	SND_PCI_QUIRK_MASK(0x104d, 0xff00, 0x9000, "Sony VAIO",
 			   ALC262_SONY_ASSAMD),
 	SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1",
-- 
cgit v1.2.3


From dbaccc0cca830efe9bb3c9e4a1cfcd6503790079 Mon Sep 17 00:00:00 2001
From: Daniel Drake <dsd@laptop.org>
Date: Mon, 9 Nov 2009 15:17:24 +0000
Subject: ALSA: hda - Tweak OLPC XO-1.5 microphone bias

Our contacts at Conexant suggested that we reduce the external
microphone bias to 50% in order to center the input signal with
the DC input range of the codec. This is because the microphone
port is DC coupled for potential use with sensors.

Signed-off-by: Daniel Drake <dsd@laptop.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 15 ++++++++++++---
 1 file changed, 12 insertions(+), 3 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 6479e65858d3..905859d4f4df 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -110,6 +110,7 @@ struct conexant_spec {
 
 	unsigned int dell_automute;
 	unsigned int port_d_mode;
+	unsigned char ext_mic_bias;
 };
 
 static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo,
@@ -1927,6 +1928,11 @@ static hda_nid_t cxt5066_adc_nids[3] = { 0x14, 0x15, 0x16 };
 static hda_nid_t cxt5066_capsrc_nids[1] = { 0x17 };
 #define CXT5066_SPDIF_OUT	0x21
 
+/* OLPC's microphone port is DC coupled for use with external sensors,
+ * therefore we use a 50% mic bias in order to center the input signal with
+ * the DC input range of the codec. */
+#define CXT5066_OLPC_EXT_MIC_BIAS PIN_VREF50
+
 static struct hda_channel_mode cxt5066_modes[1] = {
 	{ 2, NULL },
 };
@@ -1980,9 +1986,10 @@ static int cxt5066_hp_master_sw_put(struct snd_kcontrol *kcontrol,
 /* toggle input of built-in and mic jack appropriately */
 static void cxt5066_automic(struct hda_codec *codec)
 {
-	static struct hda_verb ext_mic_present[] = {
+	struct conexant_spec *spec = codec->spec;
+	struct hda_verb ext_mic_present[] = {
 		/* enable external mic, port B */
-		{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+		{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, spec->ext_mic_bias},
 
 		/* switch to external mic input */
 		{0x17, AC_VERB_SET_CONNECT_SEL, 0},
@@ -2235,7 +2242,7 @@ static struct hda_verb cxt5066_init_verbs_olpc[] = {
 	{0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */
 
 	/* Port B: external microphone */
-	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, CXT5066_OLPC_EXT_MIC_BIAS},
 
 	/* Port C: internal microphone */
 	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
@@ -2353,6 +2360,7 @@ static int patch_cxt5066(struct hda_codec *codec)
 	spec->input_mux = &cxt5066_capture_source;
 
 	spec->port_d_mode = PIN_HP;
+	spec->ext_mic_bias = PIN_VREF80;
 
 	spec->num_init_verbs = 1;
 	spec->init_verbs[0] = cxt5066_init_verbs;
@@ -2384,6 +2392,7 @@ static int patch_cxt5066(struct hda_codec *codec)
 		spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc;
 		spec->mixers[spec->num_mixers++] = cxt5066_mixers;
 		spec->port_d_mode = 0;
+		spec->ext_mic_bias = CXT5066_OLPC_EXT_MIC_BIAS;
 
 		/* no S/PDIF out */
 		spec->multiout.dig_out_nid = 0;
-- 
cgit v1.2.3


From 4ac55982907e1d48e64feaa56be91b9b52d3714d Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 10 Nov 2009 16:08:45 +0100
Subject: ALSA: hda - Avoid quirk for HP dc5750

The present quirk for HP dc5750 seems broken and maps the pins wrongly.
Since the auto-parser works well for this device, set the default entry
to use model=auto.

Reference: Novell bnc#552154
	https://bugzilla.novell.com/show_bug.cgi?id=552154

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 9bb4f75ca436..d1ccb6eaf9fc 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6249,7 +6249,7 @@ static struct snd_pci_quirk alc260_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_ACER),
 	SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FAVORIT100),
 	SND_PCI_QUIRK(0x103c, 0x2808, "HP d5700", ALC260_HP_3013),
-	SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_HP_3013),
+	SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_AUTO), /* no quirk */
 	SND_PCI_QUIRK(0x103c, 0x3010, "HP", ALC260_HP_3013),
 	SND_PCI_QUIRK(0x103c, 0x3011, "HP", ALC260_HP_3013),
 	SND_PCI_QUIRK(0x103c, 0x3012, "HP", ALC260_HP_DC7600),
-- 
cgit v1.2.3


From 71121d9fcc494453b9311992de220abb47dde3f1 Mon Sep 17 00:00:00 2001
From: Roel Kluin <roel.kluin@gmail.com>
Date: Tue, 10 Nov 2009 20:11:55 +0100
Subject: ALSA: hda - possible read past array alc88[02]_parse_auto_config()

The test of index `i' is after the read - too late - and
unsafe: if snd_hda_get_connections() fails in the last
iteration a read beyond the array is possible.

Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 8 ++++----
 1 file changed, 4 insertions(+), 4 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index d1ccb6eaf9fc..daf6975b0c2e 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4684,9 +4684,9 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
 			spec->multiout.dig_out_nid = dig_nid;
 		else {
 			spec->multiout.slave_dig_outs = spec->slave_dig_outs;
-			spec->slave_dig_outs[i - 1] = dig_nid;
-			if (i == ARRAY_SIZE(spec->slave_dig_outs) - 1)
+			if (i >= ARRAY_SIZE(spec->slave_dig_outs) - 1)
 				break;
+			spec->slave_dig_outs[i - 1] = dig_nid;
 		}
 	}
 	if (spec->autocfg.dig_in_pin)
@@ -9813,9 +9813,9 @@ static int alc882_parse_auto_config(struct hda_codec *codec)
 			spec->multiout.dig_out_nid = dig_nid;
 		else {
 			spec->multiout.slave_dig_outs = spec->slave_dig_outs;
-			spec->slave_dig_outs[i - 1] = dig_nid;
-			if (i == ARRAY_SIZE(spec->slave_dig_outs) - 1)
+			if (i >= ARRAY_SIZE(spec->slave_dig_outs) - 1)
 				break;
+			spec->slave_dig_outs[i - 1] = dig_nid;
 		}
 	}
 	if (spec->autocfg.dig_in_pin)
-- 
cgit v1.2.3


From 46ef6ec9da420b298b1f197e445bf5b06fe01ef4 Mon Sep 17 00:00:00 2001
From: Daniel T Chen <crimsun@ubuntu.com>
Date: Wed, 11 Nov 2009 14:32:10 -0500
Subject: ALSA: hda: Use model=mb5 for MacBookPro 5,2

BugLink: https://bugs.launchpad.net/bugs/462098

Until we can look closer at the verbs, let's use ALC885_MB5 for
codec SSID 0x106b4600 to enable playback and capture for MacBookPro
5,2s.

Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 5 +++--
 1 file changed, 3 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index daf6975b0c2e..84a52efdb2d6 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8911,10 +8911,11 @@ static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC885_MBP3),
 	SND_PCI_QUIRK(0x106b, 0x3e00, "iMac 24 Aluminum", ALC885_IMAC24),
 	SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC885_MB5),
-	/* FIXME: HP jack sense seems not working for MBP 5,1, so apparently
-	 * no perfect solution yet
+	/* FIXME: HP jack sense seems not working for MBP 5,1 or 5,2,
+	 * so apparently no perfect solution yet
 	 */
 	SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC885_MB5),
+	SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC885_MB5),
 	{} /* terminator */
 };
 
-- 
cgit v1.2.3


From e2e527ae7fb07caa58f8fa8fa7e90ada0b175dd7 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 13 Nov 2009 08:28:03 +0100
Subject: ALSA: hda - Add another Nvidia HDMI codec id (10de:0005)

Found on Nvidia 9800M GTS.

Reported-by: Chris Balcum <sherl0k@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_nvhdmi.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c
index 9fb60276f5c9..6afdab09bab7 100644
--- a/sound/pci/hda/patch_nvhdmi.c
+++ b/sound/pci/hda/patch_nvhdmi.c
@@ -397,6 +397,7 @@ static int patch_nvhdmi_2ch(struct hda_codec *codec)
 static struct hda_codec_preset snd_hda_preset_nvhdmi[] = {
 	{ .id = 0x10de0002, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch },
 	{ .id = 0x10de0003, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch },
+	{ .id = 0x10de0005, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch },
 	{ .id = 0x10de0006, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch },
 	{ .id = 0x10de0007, .name = "MCP7A HDMI", .patch = patch_nvhdmi_8ch },
 	{ .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch },
@@ -406,6 +407,7 @@ static struct hda_codec_preset snd_hda_preset_nvhdmi[] = {
 
 MODULE_ALIAS("snd-hda-codec-id:10de0002");
 MODULE_ALIAS("snd-hda-codec-id:10de0003");
+MODULE_ALIAS("snd-hda-codec-id:10de0005");
 MODULE_ALIAS("snd-hda-codec-id:10de0006");
 MODULE_ALIAS("snd-hda-codec-id:10de0007");
 MODULE_ALIAS("snd-hda-codec-id:10de0067");
-- 
cgit v1.2.3


From 5e08fe570c2dbabb5015c37049eb9a451e55c890 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Sat, 14 Nov 2009 14:37:19 +0100
Subject: ALSA: ice1724 - Fix section mismatch in prodigy_hd2_resume()

Remove invlid __devinit prefix from the suspend callback.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ice1712/prodigy_hifi.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c
index c75515f5be6f..6a9fee3ee78f 100644
--- a/sound/pci/ice1712/prodigy_hifi.c
+++ b/sound/pci/ice1712/prodigy_hifi.c
@@ -1100,7 +1100,7 @@ static void ak4396_init(struct snd_ice1712 *ice)
 }
 
 #ifdef CONFIG_PM
-static int __devinit prodigy_hd2_resume(struct snd_ice1712 *ice)
+static int prodigy_hd2_resume(struct snd_ice1712 *ice)
 {
 	/* initialize ak4396 codec and restore previous mixer volumes */
 	struct prodigy_hifi_spec *spec = ice->spec;
-- 
cgit v1.2.3


From 8ef5837a47f73faee18fa7ce2f9a9eb7675be8de Mon Sep 17 00:00:00 2001
From: Daniel J Blueman <daniel.blueman@gmail.com>
Date: Sat, 14 Nov 2009 18:20:04 +0000
Subject: ALSA: hda - Dell Studio 1557 hd-audio quirk

Add the Dell Studio 15 (model 1557, Core i7) laptop to the hd-audio
quirk list, enabling audio.

Signed-off-by: Daniel J Blueman <daniel.blueman@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 8eb6508cd991..86de305fc9f2 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1590,6 +1590,8 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
 				"Dell Studio 17", STAC_DELL_M6_DMIC),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02be,
 				"Dell Studio 1555", STAC_DELL_M6_DMIC),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02bd,
+				"Dell Studio 1557", STAC_DELL_M6_DMIC),
 	{} /* terminator */
 };
 
-- 
cgit v1.2.3


From bf97402052483c125a9ea7bf13df0dd9b4134078 Mon Sep 17 00:00:00 2001
From: Dan Carpenter <error27@gmail.com>
Date: Mon, 16 Nov 2009 11:07:17 +0200
Subject: ALSA: ice1724 - make some bitfields unsigned

This is a clean up and doesn't change the behavior.

Bit fields should always be unsigned.  Otherwise pm_suspend_enabled will
be -1 when you want it to be 1.  The other bad thing is that the sparse
checker will complain 36 times if they aren't unsigned.

The other bitfields in that struct are unsigned already.

Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ice1712/ice1712.h | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h
index 9da2dae64c5b..d063149e7047 100644
--- a/sound/pci/ice1712/ice1712.h
+++ b/sound/pci/ice1712/ice1712.h
@@ -382,8 +382,8 @@ struct snd_ice1712 {
 #ifdef CONFIG_PM
 	int (*pm_suspend)(struct snd_ice1712 *);
 	int (*pm_resume)(struct snd_ice1712 *);
-	int pm_suspend_enabled:1;
-	int pm_saved_is_spdif_master:1;
+	unsigned int pm_suspend_enabled:1;
+	unsigned int pm_saved_is_spdif_master:1;
 	unsigned int pm_saved_spdif_ctrl;
 	unsigned char pm_saved_spdif_cfg;
 	unsigned int pm_saved_route;
-- 
cgit v1.2.3


From 12929baea4b29d70525f764034b3dac771dd69e5 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 17 Nov 2009 15:58:35 +0100
Subject: ALSA: hda - Fix quirk for VAIO type G

Vaio type G laptop doesn't work with the current quirk setup.
After some tests, it turned out that it should be model=auto as default.

Reported-by: Mattia Dongili <malattia@linux.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 84a52efdb2d6..70583719282b 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -11462,6 +11462,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x104d, 0x9016, "Sony VAIO", ALC262_AUTO), /* dig-only */
 	SND_PCI_QUIRK(0x104d, 0x9025, "Sony VAIO Z21MN", ALC262_TOSHIBA_S06),
 	SND_PCI_QUIRK(0x104d, 0x9035, "Sony VAIO VGN-FW170J", ALC262_AUTO),
+	SND_PCI_QUIRK(0x104d, 0x9047, "Sony VAIO Type G", ALC262_AUTO),
 	SND_PCI_QUIRK_MASK(0x104d, 0xff00, 0x9000, "Sony VAIO",
 			   ALC262_SONY_ASSAMD),
 	SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1",
-- 
cgit v1.2.3